Understanding Dual INVITE Messages in Dual-Interface SIP Configuration

Hello FreePBX Community,

I am seeking advice on a particular behavior observed in my FreePBX setup, which involves dual network interfaces (eth0 and eth1). My current configuration uses eth0 for general and local network traffic (IP:, while eth1 is dedicated to VoIP traffic with my SIP provider (IP:

Issue Observed:

When making internal calls (e.g., from extension 1000 to extension 1001), I notice two distinct SIP INVITE messages being generated:

  1. The first INVITE is sent over eth0 (IP:
  2. The second INVITE is sent over eth1 (IP:


  1. Is this behavior normal for a dual-interface setup?
  2. Could this potentially lead to any SIP traffic handling issues, especially with call setups or quality?
  3. If adjustments are needed, what would be the recommended configuration changes to ensure optimal SIP traffic handling?

Additional Context:

  • The calls are being established successfully without any quality issues.
  • The system routes SIP traffic for external calls through eth1, interfacing with the SIP provider.
  • I am concerned about the complexity and potential unforeseen issues this might cause, particularly with future network configuration changes.

I would greatly appreciate any insights, experiences, or recommendations from the community regarding this setup. If further details are required, I am happy to provide them.

Thank you in advance for your assistance!

This is not normal. At the Asterisk command prompt, does
pjsip show aor 1001
show more than one contact?

Possibly, Follow Me (or some similar setup) on 1001, that’s trying to (for example) ring a mobile along with the deskphone?

Possibly, some kind of forwarding on the 1001 device, that’s initiating a new call?

What does the bogus INVITE show (source and destination IPs, source and destination numbers)?

Thank you for your responce.
This is my outcome for pjsip show aor 1001
Contact: 1001/sip:[email protected]:57255;rinstance= 17b5c09297 Avail 26.989
The INVITE message in this context shows the following:

Update on SIP Traffic Behavior for Internal Calls

Hello again,

I have an update on my previous query regarding the dual INVITE messages in SIP traffic on my FreePBX system. I’ve noticed another peculiar behavior: when making internal calls between extensions (e.g., from extension 1000 to 1001), the caller ID displays the IP address of my external interface ( instead of the expected local network IP (

This behavior is puzzling because the external interface is dedicated to SIP traffic with my VoIP provider and shouldn’t be involved in internal calls. I’ve checked my NAT and SIP settings, but I’m still encountering this issue. Could this be related to my NAT settings, routing tables, or some misconfiguration in the ifcfg-eth0 and ifcfg-eth1 files?

Any insights or suggestions to resolve this would be greatly appreciated.

Thank you!

I have several dual NIC setups. The key to make it work is to define the correct IPs and networks on the Advanced SIP configuration page.

My System configuration

  1. Network Interfaces:
  • eth0 : Configured with IP, subnet mask, and gateway It’s set as the default route (DEFROUTE=yes)
  • eth1 : Configured with IP, subnet mask, and gateway This interface is dedicated to SIP traffic with your VoIP provider and does not have a default route (DEFROUTE=no).
  1. DNS Configuration:
  • My /etc/resolv.conf file specifies DNS servers in the following order:, (both provided by your VoIP provider), (my Router), and Google’s public DNS
  • The primary DNS server for my system is, as it’s the first one listed in resolv.conf.My second line is 192.168…3
  1. Behavioral Observations:
  • Internally, calls between extensions (e.g., from 1000 to 1001) show the caller ID as [email protected] on the phone display, which is unusual as this should typically only show the extension number (e.g., 1000).
  • SIP traffic analysis reveals dual INVITE messages for internal calls – one referencing the network and the other referencing the network.
  1. Concerns and Queries:
  • I am seeking clarification on why internal calls display full SIP URIs (including the IP) instead of just the extension numbers.
  • My extensions are registered at . Why is there the IP involved in the INVITE Messages. This is the IP of eth1 which is dedicated to Sip Traffic from my provider
    Thank you for your help

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